VoIP has become more and more popular as various mass-market services have capitalized on the expanding availability of Internet access. VoIP has been implemented in various ways using both proprietary and open protocols and standards. Examples of technologies used to implement VoIP include: H.323; IP Multimedia System (IMS); Session Initiation Protocol (SIP); and, Real-time Transport Protocol (RTP).
RTP is used extensively in VoIP communication and entertainment systems that involve streaming media, such as internet telephony, video teleconference applications, and web-based push-to-talk features. RTP was developed by the Audio-Video Transport Working Group of the Internet Engineering Task Force (IETF) and first published in 1996 as Request for Comments (RFC) 1189. This version was superseded in 2003 by RFC 3550.
While the advent of VoIP using RTP has provided many benefits, one of the drawbacks has been the ease with which third parties can intercept a VoIP transmission and record the conversation. While several standards have been developed for encryption of data flow, such as the Secure Real-time Transport Protocol (SRTP) and Media Path Key Agreement for Secure RTP (ZRTP), some VoIP providers and networks will not process encrypted data without specific knowledge of the SRTP/ZRTP/security protocols, including any potential keying and credential material. SRTP has the facilities to secure and sign the entire RTP payload, instead of just the audio payload. For example, any network infrastructure component or relay server that needs to modify the RTP header information for its own purposes must have knowledge of the session key(s) in order to modify the contents of any signed RTP header information.
Nevertheless, RTP with its associated security protocols, in conjunction with the standard User Datagram Protocol (UDP) and Internet Protocol (IP) encapsulation, exhibit the problem of adding significant overhead in terms of bandwidth consumption to the data transmissions by the parties involved in the communications. While this overhead may be capably handled by many of the newer networks available today, these transmissions may exceed the capacity of some of the existing infrastructure in some of the less-developed or rural/remote areas of the world or where a network connection is made through the use of a wireless wide area network (WWAN).
In addition to the bandwidth consumption problem, there are also service issues when RTP is used in conjunction with UDP. UDP does not guarantee the delivery, sequence, or uniqueness of any RTP payload, thus resulting in the occasional loss of audio packets. Furthermore, information in RTP headers is sometimes modified or changed when transferred among networks and servers and communication of RTP headers is not guaranteed end-to-end.
It would therefore be desirable to be able to reliably encrypt VoIP communications via RTP transmissions while minimizing or reducing the amount of overhead required for secure data transmission of media content.